1. Field of the Invention
The present invention relates generally to a voice communication processing system and, more particularly, to a voice communication processing system and method for processing a speech waveform as a digital bit stream.
2. Description of the Related Art
Digital voice communication is used in a number of applications and has been increasingly used in military communications to provide high-security transmission of speech. Voice communication systems therefore have been implemented which transmit digitized speech at 2400 bits per second over a single channel. Such a 2400 bits per second system is currently deployed with a linear predictive coder. However, a more efficient and effective (error free) data transfer rate for speech signals with similar quality as the 2400 bits per second systems, for example, 800 bits per second, is desirable.
A voice communication system which processes and transmits intelligible speech at a more efficient data rate, such as 800 bits per second, would provide a number of advantages not currently available. For example, increased tolerance to channel bit errors could be provided. Conventionally, the intelligibility of the 2400 bits per second linear predictive coder degrades quickly in the presence of bit errors during transmission. Providing a voice communication system with a data transfer rate of 800 bits per second which has similar quality of a 2400 bits per second speech signal would allow for the addition of error protection coding to be added to the 800 bits per second speech data for transmission at 2400 bits per second and would thus increase the tolerance to bit errors at existing transmission speeds.
Additionally, a more efficient data rate would allow a low probability of intercept (LPI) to be maintained. With a lower data rate for the same speech signal, speech can be transmitted over channels having a smaller bandwidth and/or each speech segment can be transmitted in a shorter period of time on a conventional 2400 bits per second channel. For this reason, a very low data rate is an indispensable element of an LPI voice system. Currently, a great deal of effort is in progress to implement LPI voice terminals.
Also, a more efficient data rate would allow for voice/data integration. Recently, voice/data integration has drawn a great deal of attention. The use of an 800 bits per second voice encoding system would allow integration of voice and data over a single 2400 bits per second channel. For example, visual aids, such as written text or drawings, could be transmitted along with the voice data to enhance communicability.
Finally, a more efficient data rate would allow for voice multiplexing or, voice/voice integration. Currently, a single voice signal can be transmitted over a 3 kHz narrowband channel. If an 800 bits per second voice processor is used, however, three independent voice signals could be multiplexed and transmitted over a single narrowband 2400 bits per second channel. This multiplexing capability would permit secure conferencing, that is, three speakers at one site could communicate with three speakers at another site. Conventionally, secure conferencing has required a conference director to moderate the traffic flow by designating which party can talk, which is not a practical solution to conferencing objectives. With voice multiplexing, however, it would become possible to transmit three individual voices independently over a single channel. As a result, all participants can hear each other, even if two people accidentally talk at the same time. The provision of a voice communication system having a more efficient data rate for a speech signal, for example, 800 bits per second, is desirable to accomplish all of the above features.